IP telephony is a collection of technologies that emulates and extends today's circuit-switched telecommunications services to operate on packet-switched data networks based on the Internet Protocol (IP) IP telephony encompasses the terms “Internet Telephony”, “voice-over-IP” (VoIP), “video-over-IP”, and “fax-over-IP”, and extends those capabilities even further to include new telecommunications applications made possible by the convergence of voice, video and data. “Voice-over-IP” (VoIP) technology enables the real-time transmission of voice signals as packetized data over “IP networks” that employ the Transmission Control Protocol (TCP), Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and Internet Protocol (IP) suite, for example.
A conventional Public Switched Telephone Network (PSTN) provides its users with dedicated, end-to-end circuit connections for the duration of each call. Based on the calling and called parties' numbers, circuits are reserved among an originating switch, any tandem switches along the route between the two ends of the call, and a terminating switch. Signaling between these PSTN switches supports basic call setup, call management, and call teardown as well as querying of databases to support advanced services such as local number portability, mobile subscriber authentication and roaming, virtual private networking, and toll-free service.
The conventional PSTN has served voice traffic well over the last 100 years, but its success has been paralleled by a rise of separate networks to support data traffic. These separate networks include, for example, the World-Wide Web which is commonly referred to as the Internet, an Intranet, a wide-area network (WAN), a local area network (LAN), an ATM, a T1 network, an E1 network, an Ethernet, a microwave network, a satellite network or the like, or a combination thereof. Clearly, use of distinct networks for voice and data represents an additional burden to service providers and an additional cost to consumers. As more and more PSTN traffic becomes data-oriented, however, the trend toward voice and data network convergence becomes stronger and stronger. Service providers, Internet service providers, and manufacturers of switching, transmission, and customer premises equipment are all participating in a significant shift of the telecommunications industry toward combined voice/video/data networking using IP.
The shift to IP telephony promises better efficiencies in the transport of voice and data, and, as a result, lower telecommunications costs to end users. Moreover, as IP telephony evolves, it will be able to match all the features of voice communications currently supported by the PSTN. Interoperability among the IP telephony products of different vendors is the first major hurdle to overcome. The real promise of IP telephony, however, will be realized with the next wave of advanced services that will begin to surpass the capabilities of the PSTN.
There are, however, some drawbacks associated with existing IP telephony systems. For example, in VoIP systems, most VoIP clients need to connect to a VoIP proxy server in order to complete a call. After the VoIP client connects to a VoIP proxy server, the overall system can provide services for the VoIP client. In conventional systems, since the VoIP proxy server is responsible for tracking the status of each connected VoIP client, on going calls and other services, the VoIP proxy server may be overloaded if too many clients attempt to connect to it. This may reduce the quality of service. This also increases the workload on the designated VoIP proxy server, even though other VoIP proxy servers in the VoIP system may have much lower workloads. This occurs even if the VoIP client connects to VoIP proxy servers connected in a conventional round robin manner. Round robin connection does not guarantee that the workload will be equally distributed among the VoIP proxy servers. In many instances, the VoIP client lacks information on the workload of each VoIP proxy server and is unable to connect to a less loaded VoIP proxy server. One attempted solution is to configure multiple proxy servers to receive a request to connect from a VoIP client. One of the proxy servers is configured as the primary proxy server. If the VoIP client cannot connect to the primary proxy server, then the VoIP client transmits a request to a secondary VoIP proxy server. The VoIP client may be configured to attempt to connect to each of the VoIP proxy servers in the VoIP proxy server group in a predefined sequence by directly transmitting the request to connect to each one. Alternatively, service providers provide a list of available servers and let the user try and pick a less busy one. This places the burden on the user who must manually select a VoIP proxy server from the list and attempt to connect to the newly selected VoIP proxy server. In this scenario, the user usually lacks information on the workload level of the VoIP proxy servers and relies on “luck” to select a less loaded VoIP proxy server with which to attempt to connect.
Therefore, there exists a strong need in the art for a system and method which automatically identifies and utilizes the VoIP proxy server with a lower workload. Such a system and method would provide the identity of the VoIP proxy server with the lower workload to the VoIP client in order to more efficiently complete the call.